Voice telephony has been known for many years. Initially, voice telephony was supported by dedicated conductors between telephones. Then, voice telephony was enabled by operators manually switching connectors to create and tear down circuits between telephones. As technology advanced, mechanical components performed the switching operations to create and tear down circuits between telephones. With advancing technology, computers and semiconductor components replaced the mechanical components to perform circuit switching duties. Networks created using this circuit-switched technology are generally known as the Public Switched Telephone Network (PSTN). Generally, the PSTN provides a circuit-switched, time-divided connection between telephones.
Packet data communications, such as those supported by the Internet, differ from circuit-switched communications. With packet data communications, a source device forms a data packet, transmits the data packet to a packet data network, and based upon a destination address, e.g., Internet Protocol (IP) address of the data packet, the packet data network passes the data packet to a destination device. As the Internet and other packet data networks grew in popularity, packet switched voice telephony was developed. One common type of packet switched voice telephony is Voice over Internet Protocol (VoIP) telephony. When VoIP telephony was first introduced, the data packet transmission latency of the Internet and of other servicing networks caused the quality of VoIP telephony to be significantly worse than that of PSTN telephony. Over time, packet data transmission latency of the Internet and of other servicing packet data networks has decreased. Now, VoIP telephony provides service quality equal to or better than PSTN telephony in many cases.
Recently developed VoIP telephony applications enable computer users to establish non-toll VoIP telephone calls across the Internet. Compared to PSTN telephony VoIP telephony of this type is significantly less expensive, particularly for overseas calls. However, only a limited number of people have a computer upon which this VoIP telephony application may be loaded and have Internet access of a quality that will support the VoIP telephony application.
The underpinning of the Next Generation Network is Internet Protocol with Multi-Protocol Layer Switching (IP/IMPLS). Thus, the traditional PSTN is becoming a network edge access point into the IP/MPLS network via PSTN/VoIP gateways, built on top of a Session Initiation Protocol (SIP), Real-Time Transport Protocol (RTP) or other internet standard. Moreover, ENUM-based directories that allow mapping of traditional phone numbers to IP Uniform Resource Identifiers (URIs) enable telecommunications service providers to provide new personal mobility solutions. People can use a single phone number and control how the call gets directed to either an IP termination point, an arbitrary telephone number, a wireless (cellular) roaming phone or potentially any of a number of alternate contact points.
Due to the range of possible outcomes that come from dialing a phone number, the calling party needs to have some way of controlling the call routing behavior. It would be beneficial for the calling party to get information regarding the called party, such as the location of the called party, convenient times to contact the called party, and/or which of a series of contact points the call should be directed to. Furthermore, the calling party needs to be made aware of any potential new charges that can incur from selecting one of the available options.
In current scenarios, it is possible to redirect phone numbers to other phone numbers (typically within one's own geography) and it is also possible to have a Global System For Mobile Communication (GSM) (or other cellular wireless) service that provides a global roaming capability, however in both cases the call is transparently routed whether to the requested number or across the visited wireless network and any charges for such a service are charged to the called party, not to the calling party.
U.S. Patent Application No. 2006/0210032 is directed to a method for communicating between a calling party and a called party. The method includes receiving information indicative of an incoming call form the calling party, accessing context information associated with the called party, and accessing a pass code provided by the calling party. The method includes disposing of the incoming call based on the pass code and the context information. The method does not allow the calling party to have discretion to select whether he wishes to disturb the called party by making the call go through or electing to leave a message or return identification to allow the called party to return the call at a later time.
It is a primary object to provide a communications system and method that provides a calling party with options and flexibility regarding the direction of a call. It is a further object to provide a communications system and method that provides a calling party with the option of making a call from a PSTN telephone to a telephone using VoIP telephone service.
Description Of the Related Art Section Disclaimer: To the extent that specific publications are discussed above in this Description of the Related Art Section, these discussions should not be taken as an admission that the discussed publications (for example, published patents) are prior art for patent law purposes. For example, some or all of the discussed publications may not be sufficiently early in time, may not reflect subject matter developed early enough in time and/or may not be sufficiently enabling so as to amount to prior art for patent law purposes. To the extent that specific publications are discussed above in this Description of the Related Art Section, they are all hereby incorporated by reference into this document in their respective entirety(ies).